Помогите прочитать лог и понять его
Добавлено: 19 авг 2014, 17:51
помогите прочитать лог
Пытаюсь изучать диалпланы написал такой:
asterisk*CLI> dialplan show zpupdaes
[ Context 'zpupdaes' created by 'pbx_config' ]
'123' => 1. Wait(5) [pbx_config]
2. Dial(sip/101) [pbx_config]
3. NoOp(${DIALSTATUS}) [pbx_config]
4. GotoIf($["${DIALSTATUS}"!="ANSWER"]?1:play) [pbx_config]
[play] 5. Playback(zpupdaes) [pbx_config]
6. Playback(zpupdaes) [pbx_config]
7. Playback(zpupdaes) [pbx_config]
8. Hangup() [pbx_config]
-= 1 extension (8 priorities) in 1 context. =-
asterisk*CLI>
При снятии трубки я не получаю проигрывания файла. Помогите понять почему. Вот лог.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called sip/101
-- SIP/101-0000001c is ringing
-- Got SIP response 480 "Temporarily not available" back from 10.138.150.34:5060
-- SIP/101-0000001c is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [123@from-internal:3] NoOp("SIP/100-0000001b", "CONGESTION") in new stack
-- Executing [123@from-internal:4] GotoIf("SIP/100-0000001b", "1?1:play") in new stack
-- Goto (from-internal,123,1)
-- Executing [123@from-internal:1] Wait("SIP/100-0000001b", "5") in new stack
-- Executing [123@from-internal:2] Dial("SIP/100-0000001b", "sip/101") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called sip/101
-- SIP/101-0000001d is ringing
-- SIP/101-0000001d answered SIP/100-0000001b
-- Executing [h@from-internal:1] Hangup("SIP/100-0000001b", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-0000001b'
== Spawn extension (from-internal, 123, 2) exited non-zero on 'SIP/100-0000001b'
asterisk*CLI>
Пытаюсь изучать диалпланы написал такой:
asterisk*CLI> dialplan show zpupdaes
[ Context 'zpupdaes' created by 'pbx_config' ]
'123' => 1. Wait(5) [pbx_config]
2. Dial(sip/101) [pbx_config]
3. NoOp(${DIALSTATUS}) [pbx_config]
4. GotoIf($["${DIALSTATUS}"!="ANSWER"]?1:play) [pbx_config]
[play] 5. Playback(zpupdaes) [pbx_config]
6. Playback(zpupdaes) [pbx_config]
7. Playback(zpupdaes) [pbx_config]
8. Hangup() [pbx_config]
-= 1 extension (8 priorities) in 1 context. =-
asterisk*CLI>
При снятии трубки я не получаю проигрывания файла. Помогите понять почему. Вот лог.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called sip/101
-- SIP/101-0000001c is ringing
-- Got SIP response 480 "Temporarily not available" back from 10.138.150.34:5060
-- SIP/101-0000001c is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [123@from-internal:3] NoOp("SIP/100-0000001b", "CONGESTION") in new stack
-- Executing [123@from-internal:4] GotoIf("SIP/100-0000001b", "1?1:play") in new stack
-- Goto (from-internal,123,1)
-- Executing [123@from-internal:1] Wait("SIP/100-0000001b", "5") in new stack
-- Executing [123@from-internal:2] Dial("SIP/100-0000001b", "sip/101") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called sip/101
-- SIP/101-0000001d is ringing
-- SIP/101-0000001d answered SIP/100-0000001b
-- Executing [h@from-internal:1] Hangup("SIP/100-0000001b", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-0000001b'
== Spawn extension (from-internal, 123, 2) exited non-zero on 'SIP/100-0000001b'
asterisk*CLI>